* You are viewing Posts Tagged ‘idea’

One-way audio when calling multiple SIP

Hi,

On one of our locations, I am having issues with one-way audio when I call
several phones with SIP/Phone_A&SIP/Phone_B&SIP/Phone_C. When I call the
phones individually, they work fine, so it’s not a volume setting on the
phone. Also this setup has worked at other locations.

Any idea’s what to look for?

Thanks in advance.

Lifetime & Replacement

On Mon, 2012-05-07 at 10:26 -0500, Russ Meyerriecks wrote:
> There were a few compatability issues with the tdm400′s pci interface chip
> and certain motherboards

Interesting idea, but it’s been running in this same server with the
same motherboard for at least two years now, and this problem started
only last week.

At this point, I am sure enough that the card has failed that I have
ordered a new base card. If I’m wrong, I will only be out return postage
and 10% restock fee.

Transcoding degradation G711iLBC

On 04/15/2012 07:26 PM, Patrick Lists wrote:
> On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
>> Is it a good idea to use asterisk transcoding from G711 to iLBC or
>> should I find out any other solution not involving transcoding (f.e.
>> using G.729 that is supported in both sides). I’m worried about voice
>> quality and trying to avoid paying for G.729 licensing.
>>
>> Anybody with experience or quantitative measurements of the voice
>> quality degradation in that scenario?
>
> The term that may interest you is “Mean Opinion Score” and iLBC is
> quite good. See http://en.wikipedia.org/wiki/Mean_opinion_score
There’s lies, damn lies and mean opinion scores. The chart on that
wikipedia page is mostly for humour value.

Regards,
Steve

Getting Mac Address on connected IP phones

I am struggling to get the mac-addresses of IP phones that are connected
to asterisk as the phone are in different VLAN with * and they were
manually configured. I want to centralize their configuration using
res_phoneprov or tftp

I have tried nmap and arp in vain.

Any idea?

Sam

Transparent Elastix 2.2 fax receiving problem

Hi all,

First off all let me apologize for my English.

My company use an Avaya PBX with one E1 and about 250 extensions
(phones, faxes alarms, visa card readers etc).
My idea was to install a transparent asterisk between TELCO and PBX.

I use an OpenVox DE210E.
After install Elastix 2.2.0 and run dahdi_genconfig, i edit
/etc/asterisk/dahdi-channels.conf changing span 1 group from 0,11 to 0
and span 2 from 0,12 to 1.

Then from gui add all the active avaya extensions as custom extension
with dial DAHDI/i2/extension-number

The phones works fine but faxes open line; trying to synchronize (the
well known sound from faxes and modems). My home fax show the other
side number for about 90 sec and then display “comm. error” and hung up.

Any idea how to fix it ?

D. Sidirokastritis
NOC HCMR-Crete
tel. 2810-337709

Asterisk 1.8.9.3 Now Available

Does anyone have an idea on when 1.8.9.3 might show up in the RPM repositories?

Thanks!

EKG